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= Features/SIP Witch Domain Telephony =

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== Summary ==

== Summary ==

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This feature provides a foundation for users to construct and participate directly in an autonomously generated public VoIP network for voice, video, and instant messaging which requires no central service provider to maintain or perform user lookup nor proprietary protocols. This is done by running GNU SIP Witch as a domain service for the SIP protocol in a manner analogous to what sendmail does for the SMTP protocol (or Jabber does with xmpp), and thereby allows users of any standard-compliant SIP softphone or device to directly peer calls to remote users using SIP uri's and DNS lookup alone. This same service allows the user's local SIP capable devices to be managed as a local phone system as well as for calling remote users. By peering calls directly, user agents which support secure media protocols, such as zfone, Twinkle, and SIP communicator, can create entirely secure peer to peer calls between users easily to communicate privately. The long-term goal is to replace solutions like Skype with entirely free software using standard protocols.

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This feature enables an entirely free software alternative to Skype using standard IETF protocols along with support for direct peer-to-peer secure communications such as possible with ZRTP capable softphones and without need for a mitigating "service provider". Instead, DNS will be used for directly resolving SIP uri's and each user or organization will construct the network directly from the bottom-up running a sipwitch service daemon to answer and route calls for their users or on individual workstations, and do so while using existing standard compliant VoIP clients such as Empathy, Twinkle, SIP Communicator, local SIP devices, etc, as they prefer. SIP Witch will shortly be announced as having a key role in the FSF effort to promote replacing Skype with free software, and this spec is consistent with that plan.

We are essentially "feature complete" from the perspective of delivering Domain Telephony functionality for Fedora 13. Immediate work on comps, on drafting an updated howto based on the new 0.7.x release, and especially on finding and resolving bugs with user deployment, can now begin.

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Deferred features: gui setup. New cli commands, siprealm and sippasswd, were introduced instead. gui, and even a dbus applet, may be defined for F14. NAT through ipfw was deferred to get NAT functionality immediately testable. We are only offering generic RTP proxy for F13.

== Detailed Description ==

== Detailed Description ==

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This is described as a feature because the goal is ultimately to setup, deploy, and manage sipwitch (or general VoIP) services in a manner consistent with other core Fedora services. Long term goals to further this can include integrating phone service management with user account creation, use of auto-activating SIP user agents for SIP uri's (as a selectable preferred GNOME application much like email and web browser), address book integration to VoIP, and many other areas that touch widely upon Fedora and the overall desktop user experience. A much less ambitious starting point is proposed in this limited feature spec for the F13 release timeframe. From the F13 perspective and timeframe, this is principally an infrastructure feature to put things in place for later enabling the full user experience.

+

GNU SIP Witch is to SIP much like what Jabber is for xmpp. This feature is to setup, deploy, and manage sipwitch (or general VoIP) services in a manner consistent with other core Fedora services as well as to create a free public communication network anyone can participate in directly and securely to replace Skype. Long term goals to further VoIP integration in Fedora can include integrating phone service management with user account creation, use of auto-activating SIP user agents for SIP uri's (as a selectable preferred GNOME application much like email and web browser), address book integration to VoIP, and many other areas that touch widely upon Fedora and the overall desktop user experience. This particular spec however focuses on a more limited feature for the F13 release timeframe that is also a FSF priority initiative.

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This feature essentially requires a user to setup and deploy a GNU sipwitch server and connect it to manage one or more local SIP user agents, devices, or existing services. This can be be done directly on individual workstations, on a public facing server acting as a SIP agent for an entire domain and organization, or some combination of both. SIP Witch also will mitigate NAT issues on behalf of local SIP clients using the service to call remote users, thereby simplifying the deployment of such services. The goal is to be able to setup and deploy sipwitch and softphones to use it with much the same simplicity that one does for Skype, as well as to offer more extensive features when using sipwitch as a local SIP phone system for SIP devices.

+

This feature essentially requires a user to be able to easily setup and deploy a GNU sipwitch server and connect it to manage one or more local SIP user agents, devices, or existing services. This can be be done directly on individual workstations, on a public facing server acting as a SIP agent for an entire domain and organization, or some combination of both. SIP Witch also will mitigate NAT issues on behalf of local SIP clients using the service to call remote users, thereby simplifying the deployment of such services. The goal is to be able to setup and deploy sipwitch and softphones to use it with much the same simplicity that one does for Skype, as well as to offer more extensive features when using sipwitch as a local SIP phone system for SIP devices.

Since SIP Witch only mitigates SIP and will soon offer media packet forward RTP for SIP devices behind a NAT, while still establishing direct peer-to-peer communication between endpoints, it has very little overhead and no issues with patent encumbered codecs. Because peer to peer media connections are used between endpoints, sipwitch can operate directly with, manage, and scale "Social Key Verification" systems such as ZRTP. This offers the ability to use verifiable high-grade end-to-end media encryption to easily establish and maintain "secure" VoIP calls with remote users, something not possible with solutions like Asterisk which do not offer media peering at connection and require central decryption. This means trustworthy intercept-free calling can become possible for larger organizations in a very simple way.

Since SIP Witch only mitigates SIP and will soon offer media packet forward RTP for SIP devices behind a NAT, while still establishing direct peer-to-peer communication between endpoints, it has very little overhead and no issues with patent encumbered codecs. Because peer to peer media connections are used between endpoints, sipwitch can operate directly with, manage, and scale "Social Key Verification" systems such as ZRTP. This offers the ability to use verifiable high-grade end-to-end media encryption to easily establish and maintain "secure" VoIP calls with remote users, something not possible with solutions like Asterisk which do not offer media peering at connection and require central decryption. This means trustworthy intercept-free calling can become possible for larger organizations in a very simple way.

Part of the focus in F13 is on completing sipwitch NAT and then enabling a simple means for users to minimally configure and use the service. This goal will be served by a simple system "admin" application (in GNOME menus under system->administration) which will offer a form with basic questions such as the "calling domain", information about publicly appearing address for NAT, and the basic dialing plan for local SIP user agents or devices. It will include a simple tool to transform a local user account into a SIP user. A patch may also be added to the Twinkle SIP softphone to add a local SIP service such as sipwitch as a wizard "profile".

Part of the focus in F13 is on completing sipwitch NAT and then enabling a simple means for users to minimally configure and use the service. This goal will be served by a simple system "admin" application (in GNOME menus under system->administration) which will offer a form with basic questions such as the "calling domain", information about publicly appearing address for NAT, and the basic dialing plan for local SIP user agents or devices. It will include a simple tool to transform a local user account into a SIP user. A patch may also be added to the Twinkle SIP softphone to add a local SIP service such as sipwitch as a wizard "profile".

+

+

Since SIP Witch tethers SIP clients and intercommunications entirely through standard SIP protocols, SIP Witch can also be used in conjunction with and to enhance existing IP-PBX solutions such as Asterisk. There is also a sipwitch plugin that allows one to use SIP Witch to manage calls peer-to-peer to "secure" destinations using secure extension numbers while cross-registering sipwitch managed extensions to an insecure IP-PBX such as asterisk so that people can place calls to and receive calls from insecure destinations as well. This use case is outside of this initial spec, but will likely be elaborated on post F13.

== Benefit to Fedora ==

== Benefit to Fedora ==

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Further enabling Fedora users to more easily communicate and collaborate realtime in voice and video worldwide in both freedom and as desired privately, and without the need of mitigating service providers. Enabling any organization and enterprise to deploy secure scalable realtime VoIP networks using Fedora whether for public or private use. Long term goals include making secure VoIP and user-controlled telephony services a much more natural part of the overall user experience and a standard public infrastructure. Finally, as a back-end infrastructure, this feature is very naturally complimentary to Empathy as a means for users to communicate by empowering the community to create it's own messaging and communication infrastructure directly rather than depending on specific back-end providers like Google, MSN, Yahoo, etc.

+

Further enabling Fedora users to more easily communicate and collaborate realtime in voice and video worldwide in both freedom and as desired privately, and without the need of mitigating service providers. Enabling any organization and enterprise to deploy secure scalable realtime VoIP networks using Fedora whether for public or private use. Finally, as a back-end infrastructure, this feature is very naturally complimentary to Empathy as a means for users to communicate by empowering the community to create it's own messaging and communication infrastructure directly rather than depending on specific back-end providers like Google, MSN, Yahoo, etc.

== Scope ==

== Scope ==

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What is already done:

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What was already done (prior to F13 development):

* sipwitch packaged for fedora

* sipwitch packaged for fedora

* plugable architecture

* plugable architecture

* server interfaces for statistics and running state

* server interfaces for statistics and running state

* xml based configuration files

* xml based configuration files

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What was completed for F13 "SIPWitch Domain Calling" feature:

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* rework to standardize user behavior and make all account types profilable

* auto-configuration with user accounts added to @sipusers group or default 1000 uid

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What is no longer required:

* swig interfaces to server for perl and python.

* swig interfaces to server for perl and python.

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What is minimally required:

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What has been deferred to F14:

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* sipwitch packet forward (0.6 release planned)

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* password sync to digest through pam stack.

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* something basic to easily create a configuration file

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* gui admin page to set realm, passwd's, etc

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* documentation explaining how to use above

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* ipfw based NAT functionality

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What is desired (maybe F14):

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What is desired (maybe after F14):

* mysql or sqlite plugin to read local extension info from a database

* mysql or sqlite plugin to read local extension info from a database

* call detail collection saved into said database

* call detail collection saved into said database

* A more complete front-end which utilizes said database

* A more complete front-end which utilizes said database

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* A python tray application that can get server stats via xmlrpc

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* A python tray application that can get server stats via xmlrpc or dbus

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* Documentation for setting up a site

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* group calling and ACD functionality

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* feature group calling

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* user setable speed dial tables

== How To Test ==

== How To Test ==

Line 67:

Line 82:

Dial extension number of test client. You should receive busy.

Dial extension number of test client. You should receive busy.

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Dial an unknown extension number like 999. You should receive not found.

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Dial an unknown extension number like 777. You should receive not found.

=== test domain configuration ===

=== test domain configuration ===

Line 75:

Line 90:

== User Experience ==

== User Experience ==

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If the sipwitch daemon crashes, all active calls still remain active. Since packet filtering is used to form proxy forward, even calls behind NAT will continue.

+

If the sipwitch daemon crashes, all active calls still remain active. Since packet filtering is used to form proxy forward, even calls behind NAT will continue (NOTE: NAT through ipfwadm has been deferred to F14, hence calls through NAT will die if the server dies in F13).

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+

There is also a user experience associated with an admin application. This user experience is offered through siprealm and sippasswd utilities in F13.

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There is also a user experience associated with an admin application.

== Dependencies ==

== Dependencies ==

ucommon

ucommon

Line 98:

Line 114:

<!-- When your feature page is completed and ready for review -->

<!-- When your feature page is completed and ready for review -->

<!-- remove Category:FeaturePageIncomplete and change it to Category:FeatureReadyForWrangler -->

<!-- remove Category:FeaturePageIncomplete and change it to Category:FeatureReadyForWrangler -->

This feature enables an entirely free software alternative to Skype using standard IETF protocols along with support for direct peer-to-peer secure communications such as possible with ZRTP capable softphones and without need for a mitigating "service provider". Instead, DNS will be used for directly resolving SIP uri's and each user or organization will construct the network directly from the bottom-up running a sipwitch service daemon to answer and route calls for their users or on individual workstations, and do so while using existing standard compliant VoIP clients such as Empathy, Twinkle, SIP Communicator, local SIP devices, etc, as they prefer. SIP Witch will shortly be announced as having a key role in the FSF effort to promote replacing Skype with free software, and this spec is consistent with that plan.

We are essentially "feature complete" from the perspective of delivering Domain Telephony functionality for Fedora 13. Immediate work on comps, on drafting an updated howto based on the new 0.7.x release, and especially on finding and resolving bugs with user deployment, can now begin.

Deferred features: gui setup. New cli commands, siprealm and sippasswd, were introduced instead. gui, and even a dbus applet, may be defined for F14. NAT through ipfw was deferred to get NAT functionality immediately testable. We are only offering generic RTP proxy for F13.

GNU SIP Witch is to SIP much like what Jabber is for xmpp. This feature is to setup, deploy, and manage sipwitch (or general VoIP) services in a manner consistent with other core Fedora services as well as to create a free public communication network anyone can participate in directly and securely to replace Skype. Long term goals to further VoIP integration in Fedora can include integrating phone service management with user account creation, use of auto-activating SIP user agents for SIP uri's (as a selectable preferred GNOME application much like email and web browser), address book integration to VoIP, and many other areas that touch widely upon Fedora and the overall desktop user experience. This particular spec however focuses on a more limited feature for the F13 release timeframe that is also a FSF priority initiative.

This feature essentially requires a user to be able to easily setup and deploy a GNU sipwitch server and connect it to manage one or more local SIP user agents, devices, or existing services. This can be be done directly on individual workstations, on a public facing server acting as a SIP agent for an entire domain and organization, or some combination of both. SIP Witch also will mitigate NAT issues on behalf of local SIP clients using the service to call remote users, thereby simplifying the deployment of such services. The goal is to be able to setup and deploy sipwitch and softphones to use it with much the same simplicity that one does for Skype, as well as to offer more extensive features when using sipwitch as a local SIP phone system for SIP devices.

Since SIP Witch only mitigates SIP and will soon offer media packet forward RTP for SIP devices behind a NAT, while still establishing direct peer-to-peer communication between endpoints, it has very little overhead and no issues with patent encumbered codecs. Because peer to peer media connections are used between endpoints, sipwitch can operate directly with, manage, and scale "Social Key Verification" systems such as ZRTP. This offers the ability to use verifiable high-grade end-to-end media encryption to easily establish and maintain "secure" VoIP calls with remote users, something not possible with solutions like Asterisk which do not offer media peering at connection and require central decryption. This means trustworthy intercept-free calling can become possible for larger organizations in a very simple way.

Part of the focus in F13 is on completing sipwitch NAT and then enabling a simple means for users to minimally configure and use the service. This goal will be served by a simple system "admin" application (in GNOME menus under system->administration) which will offer a form with basic questions such as the "calling domain", information about publicly appearing address for NAT, and the basic dialing plan for local SIP user agents or devices. It will include a simple tool to transform a local user account into a SIP user. A patch may also be added to the Twinkle SIP softphone to add a local SIP service such as sipwitch as a wizard "profile".

Since SIP Witch tethers SIP clients and intercommunications entirely through standard SIP protocols, SIP Witch can also be used in conjunction with and to enhance existing IP-PBX solutions such as Asterisk. There is also a sipwitch plugin that allows one to use SIP Witch to manage calls peer-to-peer to "secure" destinations using secure extension numbers while cross-registering sipwitch managed extensions to an insecure IP-PBX such as asterisk so that people can place calls to and receive calls from insecure destinations as well. This use case is outside of this initial spec, but will likely be elaborated on post F13.

Further enabling Fedora users to more easily communicate and collaborate realtime in voice and video worldwide in both freedom and as desired privately, and without the need of mitigating service providers. Enabling any organization and enterprise to deploy secure scalable realtime VoIP networks using Fedora whether for public or private use. Finally, as a back-end infrastructure, this feature is very naturally complimentary to Empathy as a means for users to communicate by empowering the community to create it's own messaging and communication infrastructure directly rather than depending on specific back-end providers like Google, MSN, Yahoo, etc.

If the sipwitch daemon crashes, all active calls still remain active. Since packet filtering is used to form proxy forward, even calls behind NAT will continue (NOTE: NAT through ipfwadm has been deferred to F14, hence calls through NAT will die if the server dies in F13).

There is also a user experience associated with an admin application. This user experience is offered through siprealm and sippasswd utilities in F13.

The initial release of SIP Witch Domain Telephony will allow you to create and deploy scalable secure VoIP solutions both for managing a local SIP based telephone system and to call remote users over the public Internet without the need of a service provider or central directory service. This offers the freedom to organize and communicate freely and securely, and also free as in cost, too!